Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
On Feb 23, 2008, at 06:52 , Yehavi Bourvine +972-8-9489444 wrote: Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Yes, this is very true. Keeping WLAN active to stay connected to the SIP server means atrocious battery life. At least on my E60. At this point I get maybe 30 hours out of a charge when I use 30-60 minutes speaking time on the phone. jens ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Fons van der Beek wrote: I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? A simple way for you to test your indications.conf as far as the ringing goes is something like this: exten = s,1,Answer exten = s,n,PlayTones(ring) exten = s,n,Wait(30) exten = s,n,Hangup That should pick up the line and then play your locale's ring tone for 30 seconds before hanging up. If you hear ringing then indications.conf is fine, otherwise you have confirmed that there is a problem somewhere. This will have nothing to do with your carrier as the sounds are generated by asterisk itself as audio (as opposed to any kind of carrier-specific signaling). Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP peers from multiple databases
Hi, Is it possible to setup SIP peers with Asterisk Realtime from multiple databases? Thanks in advance. Ash ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voted most stable and easy to use phone?
shadowym wrote: I guess someone has to say it. Have you considered Aastra? You can argue about quality/features/functionality but I have set up both and the Aastra are definitely easier to configure and they reboot quicker. Nobody ever complains about the quality of sound or speakerphone on them either. I have to agree with you. I have deployed several 480i, 480i CT, and 9112i phones and have had a great success. At first I didn't care for them too much but since their original launch the firmware has improved vastly and they are quite nice units with a real phone feel. Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
I guess we are back to the fundamental problem: no asterisk generated sounds on the external call After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Sip show peers for the other end shows: * Name : sip.xs4all.nl Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : default Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 0475769XXX FromDomain : sip.xs4all.nl Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sip.xs4all.nl Addr-IP : 82.101.XX.XX Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 0475769XXX SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : Trevor Peirce schreef: Fons van der Beek wrote: I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? A simple way for you to test your indications.conf as far as the ringing goes is something like this: exten = s,1,Answer exten = s,n,PlayTones(ring) exten = s,n,Wait(30) exten = s,n,Hangup That should pick up the line and then play your locale's ring tone for 30 seconds before hanging up. If you hear ringing then indications.conf is fine, otherwise you have confirmed that there is a problem somewhere. This will have nothing to do with your carrier as the sounds are generated by asterisk itself as audio (as opposed to any kind of carrier-specific signaling). Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
While the call is progressing sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 82.101.62.XX 0475769XXX 14151-EX-29 00101/703757593 0x4 (ulaw) No Rx: ACK 82.101.62.XX 0475769XXX 6ec6f62d57d 00103/0 0x0 (nothing)No Codec=Ulaw, still no ringing Fons van der Beek schreef: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Sip show peers for the other end shows: * Name : sip.xs4all.nl Secret : Set MD5Secret: Not set Context : default Subscr.Cont. : default Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 0475769XXX FromDomain : sip.xs4all.nl Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : MaxCallBR: 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: Yes Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sip.xs4all.nl Addr-IP : 82.101.XX.XX Port 5060 Defaddr-IP : 0.0.0.0 Port 0 Def. Username: 0475769XXX SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (ulaw:20,g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : Trevor Peirce schreef: Fons van der Beek wrote: I've overwritten the indications.conf with the one from the sourcecode, stil no luck Perhaps somebody knows what the correct value for indications.conf is when using the dutch xs4all as sip carrier?? A simple way for you to test your indications.conf as far as the ringing goes is something like this: exten = s,1,Answer exten = s,n,PlayTones(ring) exten = s,n,Wait(30) exten = s,n,Hangup That should pick up the line and then play your locale's ring tone for 30 seconds before hanging up. If you hear ringing then indications.conf is fine, otherwise you have confirmed that there is a problem somewhere. This will have nothing to do with your carrier as the sounds are generated by asterisk itself as audio (as opposed to any kind of carrier-specific signaling). Trevor Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NOKIA E series Phone for SIP-VOIP calling
Jens Vagelpohl wrote: On Feb 23, 2008, at 06:52 , Yehavi Bourvine +972-8-9489444 wrote: Hello, I've one nokia E65 that works very well with my asterisk box. The people here don't let me even try it as they are afraid it will consume the battery more than when it is used the usual way. Is this true? Yes, this is very true. Keeping WLAN active to stay connected to the SIP server means atrocious battery life. At least on my E60. At this point I get maybe 30 hours out of a charge when I use 30-60 minutes speaking time on the phone. I get roughly the same on the bosses e60 and very marginally more on my e61. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
Vieri wrote: What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's REGISTERED here - if it's not registered here then try to look it up on *2 and establish the call there I tried to use DUNDi on my local servers but I can't seem to make it work. Most howtos out there explain the use of DUNDi when the extension ranges do not overlap. So in my case where both *1 and *2 have the same local extension range 4XXX, can I go the DUNDi route or should I stop bashing my head on that and explore another solution? If someone has configured a similar system then I'd greatly appreciate some tips. I read a few dundi docs like http://www.voip-info.org/wiki-DUNDi. Thanks Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have you tried placing the sip registrations in a db using realtime? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Fons van der Beek wrote: After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Alright, well let's see what ring actually is set to for your system. Let's see this from the command line: cat /etc/asterisk/indications.conf | grep country= And this from asterisk: show indications XX (where XX is your locale, of course). -- Real CNAM data for incoming Caller ID @ www.cnam.info ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it /can/ be used at the same time as any other codec - just that only /one/ codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suggestions for reliable DID provider for Canada, USA and Europe
I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Will the built-in T.38 support eliminate the need for spandsp? I'm curious how this will affect iaxmodem. MD _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Hillis Sent: Saturday, February 23, 2008 7:12 AM To: Asterisk Users List Subject: Re: [asterisk-users] FXO Cards - T38 T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it can be used at the same time as any other codec - just that only one codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Tnx for your support Trevor!! cat /etc/asterisk/indications.conf | grep country= country=nl ; default location show indications nl Country Indication PlayList = nl ringcadence 1000,4000 nl dial425 nl busy425/500,0/500 nl ring425/1000,0/4000 nl congestion 425/250,0/250 nl callwaiting 425/500,0/9500 nl dialrecall 425/500,0/50 nl record 1400/500,0/15000 nl info950/330,1400/330,1800/330,0/1000 nl stutter 425/500,0/50 The 'show indications' command is deprecated and will be removed in a future release. Please use 'indication show' instead. But Trevor, I guess this isn't the problem, because when i call from an internal location the indication is all right Also moh works from internal SIP phones to the queue. I only have a problem when i call into my asterisk box from the outside. Trevor Peirce schreef: Fons van der Beek wrote: After implementing the described test for indications.conf The CLI outputted: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/0475769XXX-095a8488, ) in new stack -- Executing [EMAIL PROTECTED]:2] PlayTones(SIP/0475769XXX-095a8488, ring) in new stack -- Executing [EMAIL PROTECTED]:3] Wait(SIP/0475769XXX-095a8488, 30) in new stack This looks OK, but there is no sound to be heard on the other end. Alright, well let's see what ring actually is set to for your system. Let's see this from the command line: cat /etc/asterisk/indications.conf | grep country= And this from asterisk: show indications XX (where XX is your locale, of course). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
On Fri, 22 Feb 2008 19:44 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: When a call arrives I check whether the REGSERVER coloumn is the same as the local server or not. If not, then there are two options: - Pass the call via IAX to the other servers; this makes both server process the call and the audio. - Send a refer message to the caller to contact the other server. You may actually want to use a redirect message for this (e.g SIP 302 response). In any case, traversing only one server in the signaling/media path as opposed to two would generally seem more efficient. -- Raj Jain mailto:rj2807 at gmail dot com sip:rjain at iptel dot org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI / Voicemail Que
Nitesh Divecha wrote: Thanks Doug, I tried that but it didn't work either... As per Wiki http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail it I think Trevor is correct. If there is a temp greeting recorded, this will be played instead of the busy. Remove it and you should be fine. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax-to-Email - Legal Issues
Hello everyone, Some months ago there were news about J2 filing lawsuits against companies using fax-to-email technology, as they claimed it was their patent. They had also won some cases, until someone filed a counter lawsuit against them based some other grounds but again related to fax-to-email. Anybody knows what is latest in this regard? Can now fax-to-email be used without fear of being sued. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
T.38 is not a codec. A codec has one input and one output. T.38 is an interactive protocol. This, however, has nothing to do with what I said. If you use G.729 in the same asterisk as my spandsp library, you are breaking my licence conditions. Steve Rob Hillis wrote: T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it /can/ be used at the same time as any other codec - just that only /one/ codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Steve Underwood wrote: I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. The new asterisk T.38 functionality is from the Asterisk addons 1.6.0b2 version of app_fax (and a few small changes in 1.6.0b4), which I thought someone would have mentioned to you, since it does use spandsp. (Or at least the configure script checks for spandsp, I haven't actually looked at the code). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Michelle Dupuis wrote: Will the built-in T.38 support eliminate the need for spandsp? I'm curious how this will affect iaxmodem. Why on earth would you want to eliminiate spandsp? (which app_fax from asterisk addons appears to use). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Steve Underwood wrote: T.38 is not a codec. A codec has one input and one output. T.38 is an interactive protocol. This, however, has nothing to do with what I said. If you use G.729 in the same asterisk as my spandsp library, you are breaking my licence conditions. Steve I should hope it isn't, I have an old CS6220 based ~ATA here (actually I've got 2 spare now), that supports T.38 fax, and when it offers a T.38 reinvite, even if you answer it you still get the G.711 stream along with the T.38 one. I don't know if this is supposed to happen, but it is a very old implementation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
I think you are missing something. Steve means that since its in add-ons its probably a GPL addition and not compatible with the g729 licensing. A t.38 gateway involves more than origination and termination, those 2 are pretty easy and do not involve any modems, the gatewaying is the harder part. Zoa. Rob Hillis wrote: T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it /can/ be used at the same time as any other codec - just that only /one/ codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys SPA-942 Phones
On Sat, 2008-02-23 at 10:09 +1100, Rob Hillis wrote: So far I've never run into anything that's even close to the speakerphone quality of the Polycoms. There's no comparison on the speakerphone between the Linksys phones and the Polycoms - it's chalk and cheese, but by the same token that holds true for just about every other phone too. For what it's worth, some of the later firmware updates have seemed to help the speakerphone quality on the Linksys phones. You may want to check to see that you're running the latest firmware. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Wow, an answer phrased in the form of a flame... A more supportive tone might actually encourage the Asterisk userbase to grow! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Kenyon Sent: Saturday, February 23, 2008 8:22 AM To: Asterisk Users List Subject: Re: [asterisk-users] FXO Cards - T38 Michelle Dupuis wrote: Will the built-in T.38 support eliminate the need for spandsp? I'm curious how this will affect iaxmodem. Why on earth would you want to eliminiate spandsp? (which app_fax from asterisk addons appears to use). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax-to-Email - Legal Issues
Zeeshan Zakaria wrote: Hello everyone, Some months ago there were news about J2 filing lawsuits against companies using fax-to-email technology, as they claimed it was their patent. They had also won some cases, until someone filed a counter lawsuit against them based some other grounds but again related to fax-to-email. Anybody knows what is latest in this regard? Can now fax-to-email be used without fear of being sued. -- Zeeshan A Zakaria The key site to look at it www.catchcurve.com. This is the name under which J2 troll their patents. If you look at the list of companies who paid up, you'll see most of the big names in telecoms. The patents actually suck, though. I didn't find a single patent of any substance. It remains to be seem whether things change in light of the recent SCOTUS ruling on obviousness. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk
You can try using the asterisk -r -x CLI command This allows you connect to the asterisk on the machine u run the command. As for APIs have I have no ideas. May be the seniors can help you. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
Jared YES That seems to be the problem! A very very long time ago I installed a X101P (an original one) and forgot about it. After issuing a modprobe ztdummy, indications on the outside line indication work as they should. After that i configured my X101P the way it should be configured! And yes! now indications are the way they should be I rebooted, I restarted asterisk and it keeps working! I want to thank everyone who helped me, Thank you all Jared Smith schreef: On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider for Canada, USA and Europe
I used TelIAX for a while and was happy with the service. I used it for testing before we connected to our PRI... http://www.teliax.com On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider forCanada, USA and Europe
I've had some serious issues with Teliax as of late with their new Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility issues. I have had zero problems with their old servers. Voicepulse has been WAY better, but no flat charges, no 729. Frankly, even my broadvoice (yikes!) connection has been significantly better, no 729. For a full Virutal PRI, I'd look at a provider that can give you the port and SIP connections, like XO. I've had good success with XO's product. -Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Saturday, February 23, 2008 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Suggestions for reliable DID provider forCanada, USA and Europe I used TelIAX for a while and was happy with the service. I used it for testing before we connected to our PRI... http://www.teliax.com On Feb 23, 2008, at 7:22 AM, Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent from D2 Technology, INC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Michelle Dupuis wrote: Wow, an answer phrased in the form of a flame... A more supportive tone might actually encourage the Asterisk userbase to grow! Okay, if you really want a more constructive answer. The addition to asterisk was an API change to allow app_fax from asterisk-addons to talk to asterisk. app_fax uses spandsp. Why on earth would you want to eliminiate the need for spandsp? It would involve re-doing really a lot of work, and spandsp is one of the finest pieces of coding to be associated with asterisk. Is that supportive enough? Bug ID #0011761 looks more interesting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider for Canada, USA and Europe
I'm using vitelity and junction for incoming, voipjet and voicepulse for outgoing. You can set your outgoing caller-id with either provider; you can not set your name. vitelity is the only provider I know of (other than consumer-grade providers like vonage) who provide incoming CID name. There are better solutions for that anyway. Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID provider for Canada, USA and Europe
I really like Junction Networks. They've been very good to deal with. Michael On Sat, 23 Feb 2008 11:30:35 -0600, Jay Milk wrote: I'm using vitelity and junction for incoming, voipjet and voicepulse for outgoing. You can set your outgoing caller-id with either provider; you can not set your name. vitelity is the only provider I know of (other than consumer-grade providers like vonage) who provide incoming CID name. There are better solutions for that anyway. Zeeshan Zakaria wrote: I posted the same question on asterisk-biz mailing list but didn't have much response. So I am posting it here now. I need a good, reliable and stable DID provider for USA, Canada and Europe. I prefer to have fixed monthly rates for incoming and outgoing calls and not per minute charges. Features I need to get with DIDs are: 1. my own caller ID and caller name on outbound calls 2. multiple channels per DID 3. g729 coded 4. canreinvite=yes option 5. IAX protocol Those who are already in this business, please advise me whom to go with. Is getting a virtual PRI a good solution? From their websites, they all look good so its hard to decide who is really good and will not disappear like Allo, or start giving voice quality issues. Thanks, -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] load balancing SIP extensions
--- Anthony Francis [EMAIL PROTECTED] wrote: Have you tried placing the sip registrations in a db using realtime? I'm not that sure I want to use realtime because I would then depend on the sql service never failing (I could use clustered active-active MySQL but that sounds overkill, or maybe not). I'll take a look at the pdf link of the previous post. Thanks Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dundi lookup
Pardon my ignorance but I understand that DUNDi lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given extension is served by some host, ie. if it's present in its dialplan. It does not say if it's registered or not. Is this correct? Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dundi lookup
On Sat, 2008-02-23 at 12:20 -0800, Vieri wrote: Pardon my ignorance but I understand that DUNDi lookups (*CLI dundi lookup [EMAIL PROTECTED]) reveal if a given extension is served by some host, ie. if it's present in its dialplan. It does not say if it's registered or not. Is this correct? That is correct. a DUNDi lookup will only tell you if that extension (or a patern match that happens to match that extension) exists in the advertising context on the DUNDi server. However, when used in conjunction with the regexten (and corresponding regcontext) settings available in both iax.conf and sip.conf, it effectively allows you to route calls to the host on which a device has registered. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
I must have started reading this thread after you reported that you actually had an AUDIO problem rather than a RINGBACK problem. The issue you experienced is a common one. Someday I hope Digium fixes that bug/design flaw. Fons van der Beek wrote: Jared YES That seems to be the problem! A very very long time ago I installed a X101P (an original one) and forgot about it. After issuing a modprobe ztdummy, indications on the outside line indication work as they should. After that i configured my X101P the way it should be configured! And yes! now indications are the way they should be I rebooted, I restarted asterisk and it keeps working! I want to thank everyone who helped me, Thank you all Jared Smith schreef: On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some dialplan help
I'm hoping someone can give me a little dialplan assistance. Here is my
scenario...
I currently have an ATT T1 connected to a Nortel Optn 11. I recently
purchased a Rhino system with a Rhino dual T1 card. What I want to do is
insert the Rhino box between the CO and the Nortel on the T1 so I can
start migrating users over to the Asterisk system in the near future.
But, in the meantime, I basically need to take all the calls that come
into the Rhino box on the first T1 (Zap/g0) just go back out on the
second T1 (Zap/g1). We have a number of DIDs that come in on that T1 and
I need them all transparently bridged for the time being.
We are running the latest Trix install on the Rhino box and I've setup
an inbound route for Zap/g0 that will dump into a custom context
(custom-nortel,s,1) Where I need the help is on the custom context part.
I'm not that strong with the Dial command and I want to make sure I get
it right because I have a very limited time to cut in the physical
insertion and I can't spend the time debugging it when it goes live.
Right now, I literally started with this:
[custom-nortel]
;This custom extension will take calls and put them on the outbound
trunk to the Nortel
exten = s,1,NoOp()
exten = s,n,Set(DIALDIGITS = ${EXTEN}) ;This should put the DID info
into DIALDIGITS
exten = s,n,Dial(Zap/1/${DIALDIGITS},,gjo) ;Dial the Nortel with
the same DID that we were called with
I know I've still got to handle voicemail and such but my main question
is if this will do what I'm looking for? Anyone done something like this
before that would have some insight?
Thanks,
Jason
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need some dialplan help
Gleim, Jason wrote:
second T1 (Zap/g1). We have a number of DIDs that come in on that T1 and
I need them all transparently bridged for the time being.
[custom-nortel]
exten = _N.,1,Dial(Zap/g1/${EXTEN})
Anything that comes in will go right back out again.
Best regards,
Trevor Peirce
--
Real CNAM data for incoming Caller ID @ www.cnam.info
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold
but your support was superior Eric! tnx for your help! Eric Wieling schreef: I must have started reading this thread after you reported that you actually had an AUDIO problem rather than a RINGBACK problem. The issue you experienced is a common one. Someday I hope Digium fixes that bug/design flaw. Fons van der Beek wrote: Jared YES That seems to be the problem! A very very long time ago I installed a X101P (an original one) and forgot about it. After issuing a modprobe ztdummy, indications on the outside line indication work as they should. After that i configured my X101P the way it should be configured! And yes! now indications are the way they should be I rebooted, I restarted asterisk and it keeps working! I want to thank everyone who helped me, Thank you all Jared Smith schreef: On Sat, 2008-02-23 at 10:28 +0100, Fons van der Beek wrote: I guess we are back to the fundamental problem: no asterisk generated sounds on the external call Do you have any T1/E1 cards in your system that aren't configured? If a zaptel card isn't taking interrupts, that would cause this same type of problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Facing problem in installing asterisk-addons
Facing problem in installing asterisk-addonsThis was fixed in the latest release of the add ons. Subject: [asterisk-users] Facing problem in installing asterisk-addons Hi, I have installed GNU gatekeeper. Then I am trying to install asterisk addons. I gave make and then make clean. I worked properly. Then I gave make install. It gave following error. make[1]: Entering directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 Please help me in understanding the solution for this. Thanking you, Regards, Preeta P Please do not print this email unless it is absolutely necessary. Spread environmental awareness. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
About the only reason for eliminating SpanDSP is compatibility with the GPL license. Remember that /any/ feature added to the free version of Asterisk is going to be added to ABE as well - ergo, the licensing of any libraries required need to be compatible with a /non/-open source license. Thomas Kenyon wrote: Michelle Dupuis wrote: Wow, an answer phrased in the form of a flame... A more supportive tone might actually encourage the Asterisk userbase to grow! Okay, if you really want a more constructive answer. The addition to asterisk was an API change to allow app_fax from asterisk-addons to talk to asterisk. app_fax uses spandsp. Why on earth would you want to eliminiate the need for spandsp? It would involve re-doing really a lot of work, and spandsp is one of the finest pieces of coding to be associated with asterisk. Is that supportive enough? Bug ID #0011761 looks more interesting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
T.38 is for all intents and purposes a codec. It's purpose is to re-encode a fax transmission as a data stream to be re-assembled at the other end as if it were a fax call. Seems to me to be pretty close to the definition of a codec to me. Your original comment was that you cannot use T.38 and G.729 in Asterisk at the same time. On a technical level, this is /not/ true, especially if the T.38 implementation does not rely on SpanDSP. (whether or not such an implementation exists is another question) Breaking license conditions is a separate issue altogether. You also appear to have answered another one of your questions on this forum to someone else (why on earth would you want to remove SpanDSP as a dependency?) by telling us that you can't run G.729 at the same time as T.38. I'm also curious as to why you assert that using G.729 in Asterisk (/not/ ABE) at the same time as a T.38 implementation that relies on SpanDSP since these are two completely separate plugins that are installed and acquired separately. That's almost like asserting that you can't run any commercial X application if you've installed my XYZ web browser on the same machine. Just because they use a common software base (X in this instance) /doesn't/ mean that you're violating the GPL by running non commercial software on the same machine. A more meaningful interpretation of the GPL would be that you either can or can't run a T.38 implementation with Asterisk /full stop/. Either the license is compatible, or it isn't. Trying to force any other interpretation on people will end up with you being dismissed as an extremist. Steve Underwood wrote: T.38 is not a codec. A codec has one input and one output. T.38 is an interactive protocol. This, however, has nothing to do with what I said. If you use G.729 in the same asterisk as my spandsp library, you are breaking my licence conditions. Steve Rob Hillis wrote: T.38 is a codec in exactly the same way that GSM or G.729 is a codec, so yes it /can/ be used at the same time as any other codec - just that only /one/ codec will be used at a time. What often happens is that the call will initially be established with a codec such as G.729 or G.711a, but once fax tones are detected the call will change codecs to T.38. According to the release notes for 1.6.0-b4... - 11873, Added core API changes to handle T.38 origination and termination (The version of app_fax in Asterisk-addons now supports this.) This should be all that is necessary to run a T.38 gateway. Steve Underwood wrote: Rob Hillis wrote: Not unless you're running CallWeaver or Asterisk 1.6.0-beta4. Asterisk has had passthrough support for T.38 for a while (somewhere in 1.4 it became available IIRC) but is currently completely incapable of terminating or encoding a fax call to T.38. I thought * was still not capable for T.38 gateway operation. Doesn't beta 4 just added T.38 termination? And, I believe it misses out some key elements of doing that properly. Note that T.38 termination is an addon, so it can't be used with, say, G.729. The only real option available at the moment is to keep one PSTN line on an ATA with an FXO port and T.38 support available and direct calls from the fax machines through to it. However, I should point out that while I believe this should be possible, I haven't actually tried it myself. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suggestions for reliable DID providerforCanada, USA and Europe
I've had some serious issues with Teliax as of late with their new Denver server. DTMF issues, IAX2 connection issues, and major latency issues. They are blaming it on 1.2 vs 1.4 asterisk compatibility issues. I have had zero problems with their old servers. Interesting... I've got several lines on Teliax that have been in place for several months and the service has been very good. Recently we connected a new system to Teliax and I've been fighting the same issues you mention. I've been told the problem is with my software since SIP seems to work fairly well but not IAX. I also found out that my system is one of the first 20 systems to connect to their new Denver server. Now I'm curious about how many others are having the same problem. John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call limits per server with Iax
If I'm using an Iax trunk between two sites what is the suggested number of calls to pass over this trunk before I run in to problems? Also is this number based on a per peer basis or all Iax calls going through the server in general? I know this will depend on the bandwidth I have between the sites, but lets take this variable out of the question and assume I have more than enough bandwidth to go around for the calls I want to process. Also to make things easy lets say I have g729 encoder cards in each server at both ends of the link. Will Iax scale well enough to allow me to pass the 92 calls or whatever the number of g729 encoder cards have for the number of encoding licenses on the cards? Thanks, tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
